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提交 60db6b53 编写于 作者: K Kailang Yang 提交者： Jaroslav Kysela
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ALSA: hda - Add support of Quanta FL1 

Added the support of Quanta FL1 with ALC269 code chip.
Also a bit space clean-ups.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>
上级 8ef355da
隐藏空白更改
内联 并排
Showing with 175 addition and 75 deletion
sound/pci/hda/patch_realtek.c
浏览文件 @ 60db6b53
......@@ -126,6 +126,7 @@ enum {
/* ALC269 models */
enum {
ALC269_BASIC,
ALC269_QUANTA_FL1,
ALC269_ASUS_EEEPC_P703,
ALC269_ASUS_EEEPC_P901,
ALC269_AUTO,
......@@ -11378,6 +11379,28 @@ static struct snd_kcontrol_new alc269_base_mixer[] = {
{ } /* end */
};
static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
/* output mixer control */
HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.info = snd_hda_mixer_amp_switch_info,
.get = snd_hda_mixer_amp_switch_get,
.put = alc268_acer_master_sw_put,
.private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
},
HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
{ }
};
/* bind volumes of both NID 0x0c and 0x0d */
static struct hda_bind_ctls alc269_epc_bind_vol = {
.ops = &snd_hda_bind_vol,
......@@ -11428,75 +11451,65 @@ static struct snd_kcontrol_new alc269_beep_mixer[] = {
{ } /* end */
};
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc269_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
static struct hda_verb alc269_quanta_fl1_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT},
{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{ }
};
/* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the
* analog-loopback mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/* toggle speaker-output according to the hp-jack state */
static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned char bits;
/*
* Set up output mixers (0x0c - 0x0e)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x680);
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_COEF_INDEX, 0x0c);
snd_hda_codec_write(codec, 0x20, 0,
AC_VERB_SET_PROC_COEF, 0x480);
}
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec)
{
unsigned int present;
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1);
}
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc269_quanta_fl1_speaker_automute(codec);
if ((res >> 26) == ALC880_MIC_EVENT)
alc269_quanta_fl1_mic_automute(codec);
}
/* set EAPD */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
{
alc269_quanta_fl1_speaker_automute(codec);
alc269_quanta_fl1_mic_automute(codec);
}
static struct hda_verb alc269_eeepc_dmic_init_verbs[] = {
{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
......@@ -11523,42 +11536,42 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = {
static void alc269_speaker_automute(struct hda_codec *codec)
{
unsigned int present;
unsigned int bits;
unsigned char bits;
present = snd_hda_codec_read(codec, 0x15, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
bits = present ? AMP_IN_MUTE(0) : 0;
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
AMP_IN_MUTE(0), bits);
AMP_IN_MUTE(0), bits);
snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
AMP_IN_MUTE(0), bits);
AMP_IN_MUTE(0), bits);
}
static void alc269_eeepc_dmic_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL,
present ? 0 : 5);
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_write(codec, 0x23, 0,
AC_VERB_SET_CONNECT_SEL, (present ? 0 : 5));
}
static void alc269_eeepc_amic_automute(struct hda_codec *codec)
{
unsigned int present;
present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0)
& AC_PINSENSE_PRESENCE;
present = snd_hda_codec_read(codec, 0x18, 0,
AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
present ? AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0));
0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
present ? AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1));
0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
}
/* unsolicited event for HP jack sensing */
static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec,
unsigned int res)
unsigned int res)
{
if ((res >> 26) == ALC880_HP_EVENT)
alc269_speaker_automute(codec);
......@@ -11590,6 +11603,76 @@ static void alc269_eeepc_amic_inithook(struct hda_codec *codec)
alc269_eeepc_amic_automute(codec);
}
/*
* generic initialization of ADC, input mixers and output mixers
*/
static struct hda_verb alc269_init_verbs[] = {
/*
* Unmute ADC0 and set the default input to mic-in
*/
{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the
* analog-loopback mixer widget
* Note: PASD motherboards uses the Line In 2 as the input for
* front panel mic (mic 2)
*/
/* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
/*
* Set up output mixers (0x0c - 0x0e)
*/
/* set vol=0 to output mixers */
{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
/* set up input amps for analog loopback */
/* Amp Indices: DAC = 0, mixer = 1 */
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
/* FIXME: use matrix-type input source selection */
/* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */
/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
/* set EAPD */
{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
{ }
};
/* add playback controls from the parsed DAC table */
static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec,
const struct auto_pin_cfg *cfg)
......@@ -11758,14 +11841,19 @@ static void alc269_auto_init(struct hda_codec *codec)
* configuration and preset
*/
static const char *alc269_models[ALC269_MODEL_LAST] = {
[ALC269_BASIC] = "basic",
[ALC269_BASIC] = "basic",
[ALC269_QUANTA_FL1] = "Quanta",
[ALC269_ASUS_EEEPC_P901] = "Asus_Epc_Dmic"
};
static struct snd_pci_quirk alc269_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1),
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
ALC269_ASUS_EEEPC_P703),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901",
ALC269_ASUS_EEEPC_P901),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101",
ALC269_ASUS_EEEPC_P901),
{}
};
......@@ -11780,6 +11868,18 @@ static struct alc_config_preset alc269_presets[] = {
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
},
[ALC269_QUANTA_FL1] = {
.mixers = { alc269_quanta_fl1_mixer },
.init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs },
.num_dacs = ARRAY_SIZE(alc269_dac_nids),
.dac_nids = alc269_dac_nids,
.hp_nid = 0x03,
.num_channel_mode = ARRAY_SIZE(alc269_modes),
.channel_mode = alc269_modes,
.input_mux = &alc269_capture_source,
.unsol_event = alc269_quanta_fl1_unsol_event,
.init_hook = alc269_quanta_fl1_init_hook,
},
[ALC269_ASUS_EEEPC_P703] = {
.mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer },
.init_verbs = { alc269_init_verbs,
......
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